[Asterisk-Dev] I Am Missing Something Somewhere Somehow!

[Charles Michaud]

Try this:

sip.conf

[101]
username=101
type=friend
secret=1234
host=192.168.10.176
context=sip
callerid="101"<101>


On Mon, 22 Nov 2004 03:45:23 +0500, Adnan Ahmed <adnan at xnet.com.pk> wrote:
> hi,
> I  am not registered my SIP Phone with Asterisk  i spend almost one day
> but find no luck.I know very well this is not  kind a problem discussed
> in this group but i try my best and all in vein so finally i am here
> hoping you ppl helping me out.I discussed this problem in
> asterisk's-users group and adding feedback from asterisk-users group my
> configs are
> 
> sip.conf
> 
> [general]
> port=5060
> bindaddr=192.168.10.195
> disallow=all
> allow=alaw
> allow=ulaw
> 
> [101]
> username=101
> type=friend
> secret=1234
> host=192.168.10.195
> context=sip
> callerid="101"<101>
> defaultip=192.168.10.176
> 
> extensions.conf
> [globals]
> [incoming]
> exten => s,1,Dial(Zap/1)
> 
> [outgoing]
> exten => _NXXXXXX,1,Dial/Zap/4/${EXTEN:0}
> exten => _0NXXXXXXXX,1,Dial,Zap/4/${EXTEN:0}
> exten => _0NXXXXXXXXX,1,Dial,Zap/4/${EXTEN:0}
> exten => _0NXXXXXXXXXX,1,Dial,Zap/4/${EXTEN:0}
> exten => 101,1,Dial,Zap/4(SIP/101)
> 
> [sip]
> exten => 101,1,Dial(SIP/101,20)
> 
> here are the console output : show no errors but also not working
> (running Asterisk in quite mode :-X ).
> *cli>sip show registry
> Host                              Username
>        Refresh State
> 
> *cli>sip show users
> Username               Secret               Authen
> Def.Context                  A/C
> 101                         12345678        md5,plaintext
> sip                                No
> 
> *cli>sip show peers
> Name/Username            Host                     Mask
>                 Port                  Status
> 101/101                        192.168.10.195    255.255.255.255
>           5060                Unmonitored
> 
> *cli>sip show channels
> Peer                User/ANR            Call ID                Seq
> (Tx/Rx)                 Lag                Jitter                Buffer
> 0 active SIP  channel(s)
>    kindly pointout my mistakes/errors and helping me out.
> I am searching wiki,google but no luck i am tried several configs but
> all in vein please please helping me out :-( .
> 
> Mike Dent wrote:
> 
> > Dont get caught by the same thing which had me ripping my hair out!
> > I had installed Fedora core 2 on a box and forgot that it had
> > installed iptables
> > firewall!
> > Type iptables -L and see if there are any rules? iptables -F will
> > flush them for the time
> > being, then try again.
> > It worked for me, wow how silly I felt!
> > Mike
> >
> >
> I am using Debian it's not working for me any other thaughts,tips
> suggestions because now i am very exhausted with this error i am looking
> almost everyplace wiki google but no luck kindly helping me out.
> 
> el Flynn wrote:
> 
> Adnan Ahmed wrote:
> 
> > hi,
> > I  am not registered my SIP Phone with Asterisk  i spend almost one
> > day  but find no luck my configs are.
> >
> >
> 
> <snip>
> 
> > *cli>sip show peers
> > Name/Username            Host
> > Mask                                  Port                  Status
> > 101/101                        192.168.10.195    255.255.255.255
> >          5060                Unmonitored
> >
> >
> 
> your "sip show peers" command shows that the phone is indeed connected
> to your Asterisk server. If you are having problems doing stuff with
> it, may I suggest you changing your dialplan to the following just to
> test things out:
> 
> [sip]
> exten => 1,1,VoicemailMain
> exten => 1,2,Hangup
> 
> then restart asterisk and dial "1" from your SIP phone. If you can
> hear the voicemail application prompts then you're okay.
> 
> flynn
> 
> Thanks In Advance .
> Adnan Ahmed.
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